Using buffered audio to overcome lapses in telephony signal

ABSTRACT

A facility for conveying first side of a voice call from a first participant to a second participant is described. Over the duration of the voice call, the facility receives the first side of the call. The facility seeks to forward the received first side of the voice call to a downstream node on a path to the second participant. The facility records the received first side of the call for at least part of the call. The facility identifies a just-ended portion of the voice call for which forwarding of the received first side of the voice call was unsuccessful. In response, the facility transmits to the downstream node the recorded first side of the voice call that coincides with the identified portion of the voice call.

BACKGROUND

In a telephone call, two people in different locations share abidirectional real-time audio link: call participant A's speech isconveyed to call participant B for participant B to hear, andparticipant B's speech is conveyed to participant A for participant A tohear. A telephone call between these participants enables them to engagein a conversation similar to one they might have if they were in thesame location, despite not being in the same location.

Various technologies support telephone calls, including Public SwitchedTelephone Networks, Primary Rate Interface, Voice Over IP, SessionInitiating Protocol, H.323, Media Gateway Control Protocol, and wirelessnetworks.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a context diagram of an environment in which the facilityoperates in some embodiments.

FIG. 2 is a block diagram showing some of the components typicallyincorporated in at least some of the computer systems and other deviceson which the facility operates.

FIG. 3 is a data flow diagram showing operation of the facility in someembodiments in a generalized environment using a continuous buffer.

FIG. 4 is a flow diagram showing a process performed by the facility insome embodiments in order to operate the lapse detector and notifier ina node downstream from a link that is protected by the facility with acontinuous buffer.

FIG. 5 is a flow diagram showing a process performed by the facility insome embodiments in order to operate the lapse remediator used by thefacility in connection with continuous buffers.

FIG. 6 is a data flow diagram showing operation of the facility in someembodiments using a continuous buffer with respect to a call in whichboth participants are using wireless phones.

FIG. 7 is a data flow diagram showing operation of the facility in someembodiments in a generalized environment where a selective buffer isused by the facility rather than a continuous buffer.

FIG. 8 is a flow diagram showing a process performed by the facility insome embodiments in order to operate the lapse detector and remediator.

FIG. 9 is a data flow diagram showing operation of the facility in someembodiments using a selective buffer for a call in which bothparticipants are using wireless phones.

DETAILED DESCRIPTION

The inventors have recognized significant disadvantages of conventionalapproaches to supporting telephone calls. Each telephony technology hasthe potential of introducing brief interruptions in a call. For example,a call supported by a wireless network may be briefly interrupted inboth directions when a wireless phone moves to a location where it nolonger has a line-of-sight to the wireless network tower to which it hasbeen connected, requiring the phone to negotiate a connection to a towerto which it does now have a line-of-sight, or when the wireless phonemoves from one cell of the wireless network to another. Voice Over IPconnections may face IP network congestion, and SIP, Public SwitchedTelephone Networks, Session Initiating Protocol, H.323, and MediaGateway Control Protocol can encounter electromagnetic interference.

The inventors have observed that these interruptions—which typicallysubstitute silence or loud, discordant noise for their calling partner'svoice—often throw conversations off course, forcing one or moreparticipants to try to understand their partner's later speech after theinterruption ends without the benefit of hearing their earlier speechduring the interruption.

To overcome these disadvantages, the inventors have conceived andreduced to practice a software and/or hardware facility for usingbuffered audio to overcome lapses in telephony signal (“the facility”).

In a telephone call, each “side” of the call has a directional pathconveying an audio signal from a microphone near one of the participantsto a speaker near the other participant. For example, for a call inwhich each of the participants is using a mobile phone, each side of thecall might have a path that begins in a speaking participant's mobilephone; traverses a radio link from the speaking participant's mobilephone to a wireless tower; the tower will transmit the data to a localdata center where core and IMS resides, the data will be then processedand traverses wired links to a switch; traverses wired links to awireless tower near the listening participant's phone; and traverses aradio link from that wireless tower to the listening participant'sphone.

In some embodiments, the facility selects one or more of a call's linksto protect with buffering. In the above example, the four wireless linksmay be regarded as particularly vulnerable to lapses, and are thereforeeach protected by the facility. For each link protected by the facilitywith buffering, the facility operates a buffer at the upstream end ofthe link. Audio received at this buffer during a lapse of the downstreamlink is recorded by the buffer, and played from the buffer through thelink when the lapse ends.

In some embodiments, the facility continues recording audionewly-received at the buffer while earlier-received audio is beingplayed from the buffer, so that none of this side of the conversation islost. In some embodiments, the facility plays buffer contents at afaster rate than they were recorded, such as twice as fast, in order to“catch up to live” more quickly. In some embodiments, the facility playsa tone or other brief sound immediately before playing buffer contents,in order to alert the listening participant that cached audio for thisside is about to be played.

In some embodiments, some or all of the buffers operated by the facilityare continuous buffers that are always recording the audio received atthe buffer, and are indexed by time of day. At the downstream end of theprotected link, a lapse detector and notifier monitors for lapses, andstores their starting time. The lapse detector and notifier is connectedto a lapse remediator at the upstream end of the protected link. As soonafter the lapse starting time as the lapse detector and notifiermonitors can communicate with the lapse remediator (The lapse mayinterrupt the ability of the lapse detector and notifier monitors tocommunicate with the lapse remediator.), the lapse detector and notifiersends a lapse message to the lapse remediator containing the lapsestarting time. In response to receiving the lapse message, the lapseremediator controls the buffer to begin playback from the starting timeindex, and the buffer continues to play these buffer contents until itcatches up to live. In some embodiments, the facility uses a circularbuffer as this continuous buffer.

In some embodiments, some or all of the buffers operated by the facilityare selective buffers that are controlled to record only during lapses.At the upstream end of the protected link, a lapse detector andremediator monitors for lapses of the link. When the lapse detector andremediator detects the beginning of a lapse, it controls the buffer tobegin recording the audio received at the buffer. When the lapsedetector and remediator detects the end of a lapse, it controls thebuffer to begin playing back the recorded audio, and the buffercontinues to play these buffer contents until it catches up to live.

In some embodiments, the facility performs voice transcription on audiostored in some or all of its buffers, and transmits the resulting textfor display to the listening participant, such as in an SMS message.

In some embodiments, the facility operates with respect to calls inwhich more than two people are simultaneously communicating. In a mannersimilar to that discussed elsewhere herein, the facility uses buffers toprotect some or all of the links used to connect the participants insuch larger calls.

By performing in some or all of the ways discussed above, the facilityprovides all of the audio spoken by each call partner to the other, evenaudio spoken during time periods when a link connecting the call lapsed.This measure of resiliency added by the facility makes calls morestraightforward, useful, time-efficient, and comfortable for theirparticipants.

Also, the facility improves the functioning of computer or otherhardware, such as by reducing the dynamic display area, processing,storage, and/or data transmission resources needed to perform a certaintask, thereby enabling the task to be permitted by less capable,capacious, and/or expensive hardware devices, and/or be performed withless latency, and/or preserving more of the conserved resources for usein performing other tasks or additional instances of the same task. Forexample, the facility can significantly reduce the duration of a call—byeliminating portions of the call during which the participants woulddiscuss the interruption caused by the lapse, and those during whichthey would repeat the information each provided during the lapse thatwas not heard—and thus the length of time for which the call occupieshardware resources. This permits the same number of calls to besupported using lower levels of hardware resources, or a greater numberof calls to be supported using the same hardware resources.

FIG. 1 is a context diagram of an environment in which the facilityoperates in some embodiments. Environment 100 includes a plurality ofcells 112 a-112 c, a plurality of wireless phones or other user devices124 a-124 b, and a communication network 110. Illustratively, the cells112 a-112 c correspond to cell sites (e.g., cellular towers) thattogether implement a 5G cellular communications network, or a wirelesscommunications network using another standard. The cells 112 a-112 c mayinclude or be in communication with base stations, radio back haulequipment, antennas, or other devices, which are not illustrated forease of discussion.

Each cell 112 a-112 c provides cellular communications over a coveragearea. The coverage area of each cell 112 a-112 c may vary depending onthe elevation antenna of the cell, the height of the antenna of the cellabove the ground, the electrical tilt of the antenna, the transmit powerutilized by the cell, or other capabilities that can be different fromone type of cell to another or from one type of hardware to another.Although embodiments are directed to 5G cellular communications,embodiments are not so limited and other types of cellularcommunications technology may also be utilized or implemented. Invarious embodiments, the cells 112 a-112 c may communicate with eachother via communication network 110. Communication network 110 includesone or more wired or wireless networks, which may include a series ofsmaller or private connected networks that carry information between thecells 112 a-112 c.

The user devices 124 a-124 c are computing devices that receive andtransmit cellular communication messages with the cells 112 a-112 c,e.g., via antennas or other means. Examples of user devices 124 a-124 cmay include, but are not limited to, mobile devices, smartphones,tablets, cellular-enabled laptop computers, or other UE or computingdevices that can communicate with a cellular network.

FIG. 2 is a block diagram showing some of the components typicallyincorporated in at least some of the computer systems and other deviceson which the facility operates. In various embodiments, these computersystems and other devices 200 can include server computer systems, cloudcomputing platforms or virtual machines in other configurations, desktopcomputer systems, laptop computer systems, netbooks, mobile phones,personal digital assistants, televisions, cameras, automobile computers,electronic media players, etc. In various embodiments, the computersystems and devices include zero or more of each of the following: aprocessor 201 for executing computer programs and/or training orapplying machine learning models, such as a CPU, GPU, TPU, NNP, FPGA, orASIC; a computer memory 202 for storing programs and data while they arebeing used, including the facility and associated data, an operatingsystem including a kernel, and device drivers; a persistent storagedevice 203, such as a hard drive or flash drive for persistently storingprograms and data; a computer-readable media drive 204, such as afloppy, CD-ROM, or DVD drive, for reading programs and data stored on acomputer-readable medium; and a network connection 205 for connectingthe computer system to other computer systems to send and/or receivedata, such as via the Internet or another network and its networkinghardware, such as switches, routers, repeaters, electrical cables andoptical fibers, light emitters and receivers, radio transmitters andreceivers, and the like. In some embodiments, devices that are wirelessphones or other devices capable of placing and conducting audio callsinclude additional components, such as a microphone for capturingspeech; a speaker for outputting speech; one or more radios and antennasfor wireless communication; data encoding, compression, encryption, androuting mechanisms; etc. While computer systems configured as describedabove are typically used to support the operation of the facility, thoseskilled in the art will appreciate that the facility may be implementedusing devices of various types and configurations, and having variouscomponents.

FIG. 3 is a data flow diagram showing operation of the facility in someembodiments in a generalized environment using a continuous buffer. Thediagram shows a call between participant A 310 and participant B 390.The call is made up of two paths. The first is from participant A'smicrophone 311, through link 313 to node 320, through link 330 to node340, and through link 334 to participant B's speaker 392. This pathconveys a first “side” of the call, in which participant A's speech isconveyed to participant B. The second depicted path corresponds to asecond side of the call, in which participant B's speech is conveyed toparticipant A. In this path, data travels from participant B'smicrophone 391 through link 393 to node 340, through link 350 to node320, and through link 316 to participant A's speaker 312. As shown inthis generalized environment, the intermediate nodes are nodes 320 and340, and the links are links 313, 330, 334, 393, 350, and 316. The nodesmay variously be mobile phones, landline phones, VOIP phones, phones ofother types, or devices of other types capable of operating as atelephony terminal and making and/or receiving phone or other voicecalls. The nodes may also be nodes that are intermediate to phones orother telephony devices, mobile base stations, routers, switches,servers, etc. Similarly, the links can be any of a wide variety of linkscapable of conveying telephony data, including wireless links, wiredlinks, optical fiber links, unguided laser or light links, etc. Thetelephony data may be encoded in any of a wide variety of ways on theselinks, including any kind and/or number of layers for compression,encryption, routing, data integrity, billing, routing security, etc. Theselection of nodes in which the facility implements lapse detector andnotifiers and lapse remediators defines the logical links that areprotected. In some embodiments, these logical links can be compoundlinks made up of two or more physical links, which may be of the same ordifferent types, and which may be joined by nodes in which the facilitydoes not implement lapse detector and notifiers or lapse remediators.

The diagram shows the facility being used to protect two links withvoice data buffering: protected link 330 from node 320 to node 340 inthe first path, and protected link 350 from node 340 to node 320 in thesecond path. With respect to protected link 330, the facility causes acontinuous buffer 324 outfitted in node 320 to continuously record theaudio data from participant A's microphone 311 received by node 320 inlink 313. In some embodiments, continuous buffer 324 and the othercontinuous buffers discussed herein are circular buffers. A circularbuffer retains the last m minutes or s seconds of received audio, anddiscards audio that is older. In various embodiments, the facilityselects this buffer residency time in a manner that balances the abilityto fully store audio for the length of expected lapses against theamount of memory consumed by the buffer and the fidelity of the audiostored in the buffer. In various embodiments, the facility uses acontinuous buffer length of five seconds, 10 seconds, 15 seconds, 20seconds, 30 seconds, 45 seconds, 60 seconds, 90 seconds, two minutes,five minutes, etc.

As participant A's audio signal continues on through link 330 towardnode 340, a lapse detector and notifier 341 of node 340 monitors receiptof audio data via link 330. If the lapse detector and notifier detects alapse—such as the failure to receive any signal from node 320, receivingdata from node 320 that contains no discernable audio, or audiodetermined by the detector to be of low quality—then the lapse detectorand notifier stores the time at which the beginning of the lapse isdetected. The lapse detector and notifier continues to monitor link 320,seeking to identify the time at which the lapse ends, such as when asignal via the link is restored, or when audio received via the link isdetermined to be of an adequate quality. At this time, the lapsedetector and notifier stores the lapse ending time, and notifies node320 of the just-ended lapse. It does this by sending an A-to-B lapsesignal 342 from the lapse detector and notifier to a lapse remediator323 of node 320. In various embodiments, the lapse signal is sent by thesame or a different means than is link 330.

When the lapse remediator receives the A-to-B lapse signal, it controlsbuffer 324 in order to begin playing the buffer's contents starting atthe lapsed beginning time contained by the lapse signal. In someembodiments, before beginning the playing of the buffer's contents, thefacility plays a distinctive tone or other short sound, or a recorded orsynthesized voice message, indicating that buffered call audio willfollow. As the buffer plays this audio to replace the correspondingaudio that was lost during the lapse, the buffer continues to recordparticipant A's audio received via link 313, without immediately passingit through to link 330. In some embodiments, the buffer plays itscontents at a higher rate than they were recorded, such as 1.25 times asfast, 1.5 times as fast, 1.75 times as fast, twice as fast, 2.5 times asfast, three times as fast, four times as fast, etc. This acceleration ofthe played-back audio permits the first side of the call to catch upwith participant A's present speech. The facility chooses a playbackrate that optimizes between catchup time and intelligibility. In orderto boost intelligibility, in some embodiments the facility processes theplayed-back audio to reduce its frequency, via techniques such asfrequency filtering, frequency reduction, pitch scaling, audio timestretching, etc. In various embodiments, the facility uses additionaltechniques in order to hasten catchup with participant A, includingdeletion or shortening of periods of silence in the played-back audio.This audio played back from buffer 324 is sent via link 330 to node 340,and through link 334 to participant B's speaker 392. When the bufferplayback catches up in the sense that the end of the recorded audio isreached—the last thing participant A said having just been replayed—thenthe lapse remediator causes audio received from the participant Amicrophone via link 313 to be routed again to link 330 toward node 340.

Bracket 331 shows the extent of protection for A-to-B audio, from atransmitter in node 320 through link 330. A similar range of protection351 is provided by the facility to the second side of the call. It canbe seen that the facility operates in an analogous way along the secondpath, conveying participant B's speech from the participant B microphone391 to the participant A speaker 312, protecting this side of the callfrom lapses that occur traveling from node 340 to node 320. Inparticular: lapse detector and notifier 321 behaves in a way analogousto lapse detector and notifier 341, sending B-to-A lapse signal 322 in away analogous to A-to-B lapse signal 342; lapse remediator 343 behavesin a way analogous to lapse remediator 323; and continuous buffer 344behaves in a way analogous to continuous buffer 324.

FIG. 4 is a flow diagram showing a process performed by the facility insome embodiments in order to operate the lapse detector and notifier ina node downstream from a link that is protected by the facility with acontinuous buffer. In act 401, the facility determines whether a lapsehas begun, in some or all of the manners discussed above in connectionwith FIG. 3 . If a lapse has not begun, then the facility continues inact 401 to continue monitoring, else the facility continues in act 402.In act 402, the facility stores the time at which the lapse began. Inact 403, the facility monitors to identify the end of the lapse in someor all of the ways discussed above in connection with FIG. 3 . If thelapse has not ended, then the facility continues in act 403 to continuemonitoring, else the facility continues in act 404. In act 404, thefacility stores the time at which the lapse ended. In act 405, thefacility notifies the upstream lapse remediator of the lapse, includingin the lapse message the lapse beginning time stored in act 402 and thelapse ending time stored in act 404. After act 405, the facilitycontinues in act 401 to monitor for the beginning of the next lapse.

Those skilled in the art will appreciate that the acts shown in FIG. 4and in each of the flow diagrams discussed below may be altered in avariety of ways. For example, the order of the acts may be rearranged;some acts may be performed in parallel; shown acts may be omitted, orother acts may be included; a shown act may be divided into subacts, ormultiple shown acts may be combined into a single act, etc.

FIG. 5 is a flow diagram showing a process performed by the facility insome embodiments in order to operate the lapse remediator used by thefacility in connection with continuous buffers. In act 501, the facilitydetermines whether a lapse notification has been received; if so, thefacility continues in act 502, else the facility continues in act 501 tocontinue monitoring. In act 502, the facility causes an alert sound tobe sent via the protected link. In act 503, the facility causes thecontents of the buffer between the beginning time included in thereceived lapse notification and catch-up time to be sent via theprotected link. As discussed above, in various embodiments, the facilityperforms various approaches to rate acceleration, silence shortening,and frequency correction to the played buffer contents before sendingthem across the protected link. After act 503, the facility continues inact 501.

FIG. 6 is a data flow diagram showing operation of the facility in someembodiments using a continuous buffer with respect to a call in whichboth participants are using wireless phones. The diagram shows awireless phone 610 used by participant A, as well as a wireless phone660 used by participant B. The diagram shows two nodes 630 and 640intermediate to the wireless phones. A link 620 to wireless phone A tonode 630 is protected, as is a link 650 from node 640 to wireless phoneB 660. In the second path, conveying the second side of the call, a link670 from wireless phone B to node 640 is protected, as is a link 680from node 630 to wireless phone A. In some embodiments, these fourprotected links each contain the wireless communications between awireless phone and a base station. In some embodiments, the intermediatenode is in the base station, such that the protected link contains onlythe wireless communications to the base station. In some embodiments,the node is somewhere in a wired network between the base station andthe other wireless phone, such as a switch, a server, etc.; in thesecases, the protected link covers the entire span from the wireless phoneto this other node. As shown, links 635 and 645 between nodes 630 and640 are not protected. It can be seen by comparing FIG. 6 to FIG. 3 thatthe facility protects these four protected links in the same way as isshown in FIG. 3 and described in connection therewith, using lapsedetector and notifiers 611, 631, 641, and 661, which can send lapsesignals 612, 632, 642, and 662; lapse remediators 613, 633, 643, and663; and continuous buffers 614, 634, 644, and 664. Those skilled in theart will appreciate that a particular side of the call can include anynumber of protected links. Also, though not shown, the two sides of asingle call may traverse different paths, which may be protectedsimilarly or differently by the facility.

FIG. 7 is a data flow diagram showing operation of the facility in someembodiments in a generalized environment where a selective buffer isused by the facility rather than a continuous buffer. When audio fromthe microphone 711 in participant A's wireless phone 710 is received bynode 720 via link 713, it is passed forward to protected link 730, andnot initially recorded by selective buffer 724. Audio from node 720 vialink 730 is received at node 740, which passes it via link 734 toparticipant B's wireless phone 790, where it is played over speaker 791.A lapse detector and remediator 723 in node 720 monitors the status ofprotected link 730 to detect any lapses. In some embodiments, the lapsedetector and remediator does this by monitoring a stream ofacknowledgment messages that would be received from node 740 if link 730was intact and operating properly. In various embodiments, the facilityuses various other approaches to monitoring for monitoring for lapses inthe lapse detector and remediator 723. When the lapse detector andremediator identifies the beginning of a lapse, it controls selectivebuffer 724 to begin recording participant A's audio received via link713. The lapse detector and remediator proceeds to monitor for the endof the present lapse. When the lapse detector and remediator detects theend of the present lapse, it directs the selective buffer to play itscomplete contents for transmission in link 730, continuing to recordparticipant A's audio received via link 713 without immediately passingit to link 730. This playing proceeds in the various manners discussedabove. When playing catches up to participant A's live audio—that is,the buffer is emptied—the lapse detector and remediator routesparticipant A's audio received via link 713 directly to link 730. It canbe seen by comparing the bottom half of the diagram to the top half thatthe facility protects protected link 750 in a similar manner. Inparticular: microphone 792 behaves analogously to microphone 711; links793, 750, and 716 behave analogously to links 713, 730, and 734,respectively; lapse detector and remediator 743 behaves analogously tolapse detector and remediator 723; and selective buffer 744 behavesanalogously to selective buffer 724. Bracket 731 shows the extent towhich the facility's protection extends for the first side of the call,as does bracket 751 with respect to second side of the call.

FIG. 8 is a flow diagram showing a process performed by the facility insome embodiments in order to operate the lapse detector and remediator.In act 801, if the downstream protected connection is operating, thenthe facility continues in act 801, else the facility continues in act802. In act 802, the facility begins recording upstream audio in theselective buffer. In act 803, if the downstream connection is operating,then the facility continues in act 804, else the facility continues inact 803. In act 804, the facility inserts the alert sound into thedownstream connection. In act 805, the facility inserts the entirecontents of the buffer. In act 806, after the entire contents of thebuffer have been inserted, the facility stops recording. After act 806,the facility continues in act 801.

FIG. 9 is a data flow diagram showing operation of the facility in someembodiments using a selective buffer for a call in which bothparticipants are using wireless phones. FIG. 9 is similar to FIG. 6 , inthe sense that nodes 930 and 940 are each at some point intermediate tothe two wireless phones 910 and 960, which may be either at the wirelessbase station that is communicating directly with the adjacent wirelessphone, or nodes in the path that are further from the correspondingwireless phone. By comparing FIG. 9 to FIG. 7 , it can be seen that thefacility protects protected links 920, 950, 970, and 980 in a similarmanner to links 730 and 750 shown in FIG. 7 and discussed above. Inparticular: links 935 and 945 pass audio in a manner analogous to links635 and 645; lapse detector and remediators 913, 933, 943, and 963operate in a manner similar to lapse detector and remediators 723 and743; and selective buffers 914, 934, 944, and 964 behave in a manneranalogous to selective buffers 724 and 744.

In various embodiments, the facility detects and remediates call lapsesof a variety of types, including some or all of the following, amongothers

1) Suddenly many SIP errors are observed on the IMS nodes.

-   -   A burst of call quality issues or call failures would start to        occur as a result of RFC 3261 defined SIP errors from different        IMS (IP Multimedia Subsystem) nodes.

2) Link flaps suddenly occur.

-   -   There will be a high level of audio/RTP/RTCP/Video quality        issues that would be noticeable as packets would be        lost/dropped.

3) Suddenly Sev 1 connectivity alarms are raised.

-   -   There will be a high level of audio/RTP/RTCP/Video quality        issues that would be noticeable as packets would be        lost/dropped.

4) Suddenly a large Spike in Memory/CPU is seen.

-   -   Due to Memory/CPU spike there will be added delay in processing        of packets at various elements resulting in out of order,        delayed packets which would result in audio quality.

5) Signaling Storm is seen.

-   -   A large amount of Signaling traffic would hit the nodes,        Signaling traffic gets priority over media as a result the media        packets would take more time to process at each hop in the        network causing noticeable audio buffering and out of order        issues.

6) K8s Worker node detects a hardware issue.

-   -   Due to Memory/CPU spike there will be added delay in processing        of packets at various elements resulting in out of order,        delayed packets which would result in audio quality.

7) K8s master node detects a connectivity/hardware issue.

-   -   There will be a high level of audio/RTP/RTCP/Video quality        issues that would be noticeable as packets would be        lost/dropped.

8) Audio Becomes bad as we move into a tunnel.

-   -   When you travel through tunnels or underground because of the        nature of Wireless communication there would be a reduction of        RSRP/RSRQ/SNR for the device, mechanisms can be put in place to        detect such behavior which causes audio buffer/choppy/out of        sync behavior.

9) Device has low battery.

-   -   Due to Memory/CPU spike there will be added delay in processing        of packets at various elements resulting in out of order,        delayed packets which would result in audio quality.

10) We are in a coverage with lot of Signal to noise ratio.

-   -   When you travel through tunnels or underground because of the        nature of Wireless communication there would be a reduction of        RSRP/RSRQ/SNR for the device, mechanisms can be put in place to        detect such behavior which causes audio buffer/choppy/out of        sync behavior.

11) We are in a Bad RSRP/RSRQ.

-   -   When you travel through tunnels or underground because of the        nature of Wireless communication there would be a reduction of        RSRP/RSRQ/SNR for the device, mechanisms can be put in place to        detect such behavior which causes audio buffer/choppy/out of        sync behavior.

12) We are in situation where we have Radio Link failure when you go toa coverage gap.

-   -   When you travel through tunnels or underground because of the        nature of Wireless communication there would be a reduction of        RSRP/RSRQ/SNR for the device, mechanisms can be put in place to        detect such behavior which causes audio buffer/choppy/out of        sync behavior.

13) Use AI/ML detections for advance notice for these faults.

-   -   Data gets collected continuously around the performance of        network elements as result AI/ML could be utilized to detect        various fault conditions in the network.

The various embodiments described above can be combined to providefurther embodiments. All of the U.S. patents, U.S. patent applicationpublications, U.S. patent applications, foreign patents, foreign patentapplications and non-patent publications referred to in thisspecification and/or listed in the Application Data Sheet areincorporated herein by reference, in their entirety. Aspects of theembodiments can be modified, if necessary to employ concepts of thevarious patents, applications and publications to provide yet furtherembodiments.

These and other changes can be made to the embodiments in light of theabove-detailed description. In general, in the following claims, theterms used should not be construed to limit the claims to the specificembodiments disclosed in the specification and the claims, but should beconstrued to include all possible embodiments along with the full scopeof equivalents to which such claims are entitled. Accordingly, theclaims are not limited by the disclosure.

1. A wireless telephony terminal, comprising: a microphone configured totransform ambient audio into audio representation data; a wirelesstelephony terminal transmitter configured to wirelessly transmit audiorepresentation data; a buffer, comprising: a buffer input configured toreceive audio representation data, a buffer memory configured to storeaudio representation data received at the buffer input, and a bufferoutput configured to provide audio representation data retrieved fromthe buffer memory; an audio representation data router configured toroute audio representation data among the microphone, the buffer input,the buffer output, and the wireless telephony terminal transmitter asfollows: for a first period, routing audio representation data from themicrophone to the wireless telephony terminal transmitter; for a secondperiod succeeding the first period, for which it is determined thataudio representation data wirelessly transmitted by the wirelesstelephony terminal transmitter will not be received by a base stationreceiver, routing audio representation data from the microphone to thebuffer input; in response to determining that audio representation datawirelessly transmitted by the wireless telephony terminal transmitterwill be received by a base station receiver, for a third periodsucceeding the second period, routing audio representation data from themicrophone to the buffer input, and routing audio representation datafrom the buffer output to the wireless telephony terminal transmitter;and for a fourth period succeeding the third period, routing audiorepresentation data from the microphone to the wireless telephonyterminal transmitter.
 2. The wireless telephony terminal of claim 1wherein the audio representation data router is configured to routeaudio representation data from the microphone to the buffer input duringthe second period in response to determining by the wireless telephonyterminal that audio representation data wirelessly transmitted by thewireless telephony terminal transmitter will not be received by a basestation receiver, and wherein the buffer memory is configured to recordonly audio representation data received at the buffer input during thesecond and third periods.
 3. The wireless telephony terminal of claim 1wherein the audio representation data router is configured to routeaudio representation data from the microphone to the buffer input for anentire duration of each call, and wherein the buffer memory isconfigured as a circular buffer to store audio representation data as itis received from the microphone for the entire duration of each call, ateach time during the duration of each call containing the audiorepresentation data received during a fixed-length interval ending atthe present time, and wherein the buffer is configured to index audiorepresentation data stored in the buffer memory by the time at which itwas received at the buffer input, the wireless telephony terminalfurther comprising a wireless telephony terminal receiver configured toreceive a lapse notification message from a base station transmitter,the lapse notification message identifying a time at which a lapse inthe audio representation data received by the base station receiver fromthe wireless telephony terminal transmitter began, and wherein thebuffer is configured to provide audio representation data at the bufferoutput beginning with audio representation data indexed with the timeidentified by the lapse notification message received by the wirelesstelephony terminal receiver.
 4. The wireless telephony terminal of claim1 wherein the buffer output is configured to provide audiorepresentation data retrieved from the buffer memory at a rate higherthan the provided audio representation data was received at the bufferinput.
 5. The wireless telephony terminal of claim 4, the wirelesstelephony terminal further comprising a pitch normalizer configured totransform the pitch of the audio representation data provided at thebuffer output to approximate the pitch of the audio representation datareceived at the buffer input.
 6. The wireless telephony terminal ofclaim 1, the buffer further comprising a notification tone generator forproviding a recorded audio playback notification tone at the bufferoutput at the beginning of the third period.
 7. A node in a telephonynetwork, the node comprising: a receiver for receiving a first stream ofaudio representation data conveying a first side of a telephony callfrom an upstream node in the telephony network; a transmitter fortransmitting a second stream of audio representation data conveying thefirst side of the telephony call to a downstream node in the telephonynetwork; a buffer, comprising: a buffer input configured to receiveaudio representation data, a buffer memory configured to store audiorepresentation data received at the buffer input, and a buffer outputconfigured to provide audio representation data retrieved from thebuffer memory; an audio representation data router configured to routeaudio representation data among the receiver, the buffer input, thebuffer output, and the transmitter as follows: for a first period,routing audio representation data from the receiver to the wirelesstelephony terminal transmitter; for a second period succeeding the firstperiod, for which it is determined that audio representation datatransmitted by the transmitter will not be received by the downstreamnode in the telephony network, routing audio representation data fromthe receiver to the buffer input; in response to determining that audiorepresentation data transmitted by the transmitter will be received bythe downstream node in the telephony network, for a third periodsucceeding the second period, routing audio representation data from thereceiver to the buffer input, and routing audio representation data fromthe buffer output to the transmitter; and for a fourth period succeedingthe third period, routing audio representation data from the receiver tothe transmitter.
 8. The node of claim 7 wherein the audio representationdata router is configured to route audio representation data from thereceiver to the buffer input during the second period in response todetermining by the node that audio representation data wirelesslytransmitted by the transmitter will not be received by the downstreamnode, and wherein the buffer memory is configured to record only audiorepresentation data received at the buffer input during the second andthird periods.
 9. The node of claim 7 wherein the audio representationdata router is configured to route audio representation data from thereceiver to the buffer input for an entire duration of the call, andwherein the buffer memory is configured as a circular buffer to storeaudio representation data as it is received from the receiver for theentire duration of each call, at each time during the duration of thecall containing the audio representation data received during afixed-length interval ending at the present time, and wherein the bufferis configured to index audio representation data stored in the buffermemory by the time at which it was received at the buffer input, thewireless telephony terminal further comprising a wireless telephonyterminal receiver configured to receive a lapse notification messagefrom the downstream node, the lapse notification message identifying atime at which a lapse in the audio representation data received by thedownstream node from the transmitter began, and wherein the buffer isconfigured to provide audio representation data at the buffer outputbeginning with audio representation data indexed with the timeidentified by the lapse notification message received by the receiver.10. The node of claim 8 wherein the buffer output is configured toprovide audio representation data retrieved from the buffer memory at arate higher than the provided audio representation data was received atthe buffer input.
 11. The node of claim 10, the wireless telephonyterminal further comprising a pitch normalizer configured to transformthe pitch of the audio representation data provided at the buffer outputto approximate the pitch of the audio representation data received atthe buffer input.
 12. The node of claim 10, the buffer furthercomprising a notification tone generator for providing a recorded audioplayback notification tone at the buffer output at the beginning of thethird period.
 13. A method in a computing system for conveying firstside of a voice call from a first participant to a second participant,the method comprising: over the duration of the voice call, receivingthe first side of the voice call; seeking to forward the received firstside of the voice call to a downstream node; recording the receivedfirst side of the voice call for at least part of the voice call;identifying a just-ended portion of the voice call for which forwardingof the received first side of the voice call was unsuccessful; and inresponse to the identifying, transmitting to the downstream node therecorded first side of the voice call that coincides with the identifiedportion of the voice call.
 14. The method of claim 13, furthercomprising: determining that the portion of the voice call for whichforwarding of the received first side of the voice call was unsuccessfulhas begun; in response to the determining, beginning the recording; andcontinuing the recording until transmitting the recorded first side ofthe voice call that coincides with the identified portion of the voicecall is complete.
 15. The method of claim 14 wherein the recordingrecords the received first side of the voice call for the entireduration of the voice call, discarding portions of the recording thatwere received more than a threshold time before the present time, themethod further comprising: indexing the recorded received first side ofthe voice call with the times across which the first side of the voicecall received; receiving from the downstream node a lapse notificationidentifying the time at which a lapse in communication between thecomputing system and the downstream node began, and wherein theidentifying identifies a portion of the voice call beginning at a timeindex corresponding to the time identified by the lapse notification,and wherein
 16. The method of claim 13, further comprising: in responseto the identifying: subjecting the recorded first side of the voice callthat coincides with the identified portion of the voice call to voicetranscription to produce a textual representation of the recorded firstside of the voice call that coincides with the identified portion of thevoice call; and transmitting the produced textual representation fordisplay to the second participant.
 17. The method of claim 13, whereinthe transmission of the recorded first side of the voice call is at arate higher than a rated which the recorded first side of the voice callwas received.
 18. The method of claim 17, further comprisingtransforming the pitch of the transmitted recorded first side of thevoice call tomorrow closely match a pitch at which the recorded firstside of the voice call was received.
 19. The method of claim 13 whereinthe receiving receives the first side of the voice call from an upstreamtelephony node.
 20. The method of claim 13 wherein the method isperformed in a wireless phone, and the receiving receives the first sideof the voice call from an analog to digital converter incorporated intothe wireless phone.